In voice conferencing systems, where the transport of audio or voice is mediated other than by a direct proximate acoustic coupling, the participants may experience an increased delay in round trip communication. Typically, in telecommunication systems, this can be of the order of 200-500 ms in each direction, what is known as ‘mouth-to-ear’ delay. This is known to have an impact on communications and functional use of such systems. ITU (ITU-T G.114 2003) sets out details of the observed impact, under different functional activities, of increased link latency. Even in simple one-to-one mediated conversations, the latency can have a substantial impact. In some cases, where long distance or adverse network conditions are experienced, typical latencies can exceed the critical threshold of 400 ms set out in (ITU-T G.114 2003).
For example, when using an IP network, typical latencies across the Atlantic network are 200 ms (http://www.verizonbusiness.com/about/network/latency/), and in addition to this, there may be additional system delays associated with buffering, central servers, jitter buffers, software systems at the end points and hardware or low level audio subsystems. Even for a well-designed system, these additional latencies may add up to 100 ms plus whatever is required for the desired robustness to network jitter.
With such latency, it may be more likely that both parties will commence speaking within the one way delay time, and then the time taken for this to be realized and one or other parties to back off. This scenario may have an impact on ‘natural turn taking’ and causes delays, stutter and inefficiency in the communications flow.
In systems that allow for many parties to communicate simultaneously, often known as voice conferencing systems, as the number of parties increases, it is sometimes efficient in bandwidth to use a central server. There may be a cost associated with this, in that all packets are typically handled by this server, with scheduling, handling, processing and output delays.